In digital transmission systems wherein digital signals are transmitted in the form of groups (packets or blocks) of bits, each group comprises data bits, destination address bits, control bits and/or flag bits delimiting the group. Due to transmission errors or node clipping, a packet or block of bits will occasionally fail to be received or contain errors. In either case, the packet or block of bits may be considered lost.
In some cases, a retransmission may be requested by the receiver, but this is not always feasible. In the particular case of voice transmission systems, operations must be performed in real time or with an extremely small delay, which precludes retransmissions. Other means for recovering or reconstructing the lost bits must therefore be devised. The properties of voice signals may be utilized as a means of facilitating this reconstruction.
The lost bits can be compensated for in the receiver by means of so-called interpolation or extrapolation processes using the bits that were normally received. Lost signal samples are calculated from correctly received samples. The calculated samples are then artificially reintroduced in the signal received by the called party.
The interpolations or extrapolations are governed by a mathematical law of variable complexity which may be linear or parabolic, for example. While imperfect, corrections made in this manner are acceptable to the human ear, provided the lost portion of signal is of short duration. However, this does not hold true if the duration of said lost portion is relatively long, as in the case of systems in which signals are transmitted in the form of packets of bits, with each packet representing, for example, a signal duration of 20 to 40 ms. Consequently, other means of compensating for lost packets must be devised.
Another method of making corrections is described in an article by N. S. Jayant and S. W. Christensen entitled "Effect of Packet Losses in Waveform Coded Speech and Improvements Due to an Odd-Even Sample-Interpolation Procedure" published in IEEE Transactions on Communications, Vol. Com. 2, No. 2, Feb. 81, pp. 101-109. Each packet or block of voice signal samples is divided at the transmitting end into two portions respectively comprising even-numbered samples and odd-numbered samples. The loss of a packet entails the loss of every second sample. The lost packets are then recovered by subjecting the correctly received samples to interpolation process. This leads to interpolating between samples rather than between packets. Unfortunately, this method increases the decoding delays, which is a disadvantage.